similar to: Asterisk log messages

Hello, One of the Sipura 2k's I'm using has a dialtone that occasionally fades to static when the user picks up the line. Are there any settings that I can check that would affect this? Regards, Christopher

Hello, I encoded a bunch of theora video files from my DVDs. I thought for a long time that mplayer was broken because it displayed the aspect ratio wrong. Then I found out that the aspect ratio I was specifying was supposed to be the pixel aspect ratio, not the video aspect ratio. (This is extremely confusing and I urge you to change the semantics of encoder_example. I also urge you to check for

I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can't specify

The Asterisk Development Team has announced the release of Asterisk 1.4.36. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.36 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.4.36. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.36 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.12 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.12 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.39 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.39 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.16. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.16 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.16. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.16 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Can someone please explain what I am doing wrong here? I only want the extensions listed in long-users to be able to access the longdistance context. If I do this, I get a congestion tone no matter what I dial. If I add a [default] context and include => longdistance, then the local callers can call the long distance number fine, which is not what I want, but I still want long-users to be

Are there any known issues with call-waiting caller-id for SIP? Caller-ID on the first call works fine, but when the second call comes in, I hear the interrupt tone, but the caller-id doesn't display anything. I have tried this with the Cisco ATA and the SPA-2000. I have also tried two different phones to verify that it wasn't something specific to the phone. Thanks, Stephen

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Has anyone succesfully integrated * with a cisco voice gateway ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/3b1ba7b3/attachment.htm

I've been attempting to calculate the predictions from a poisson glm object, along these lines: predict(foo.glm, type = "response") and predict(foo.glm, type = "response", se.fit = TRUE) foo.glm is arrived at this way: foo.glm <- glm(Insects ~ Dad * Mum + Location, offset = log(MM), family = "poisson", data = model.df) There are two

Author: krgopi Repository: /hg/zfs-crypto/gate Revision: a813fd7825c4b1d3fb282c08cdf80bc9ffa88a1a Log message: PSARC 2005/654 Nemo soft rings 6306717 For Nemo based drivers, IP can ask dls to do the fanout Files: create: usr/src/uts/common/io/dls/dls_soft_ring.c create: usr/src/uts/common/sys/dls_soft_ring.h update: usr/src/uts/common/Makefile.files update: usr/src/uts/common/inet/ip.h

We''re building for one case (CONFIG_ACPI_PCI defined) only, yet still had the other case''s code in there. Additionally there was quite a bit of pseudo-duplication between disabled(!) DMI scan and ACPI boot code. acpi_pci_disabled had only a single reader, which is off by default (i.e. must be enable on the command line), so it seems pointless to keep it. Signed-off-by: Jan

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN